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codecs: Add Codec 2 mode 2400.
ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6
This commit is contained in:
Alexander Traud
222
codecs/codec_codec2.c
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222
codecs/codec_codec2.c
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2016, Alexander Traud
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*
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* Alexander Traud <pabstraud@compuserve.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Translate between signed linear and Codec 2
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*
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* \author Alexander Traud <pabstraud@compuserve.com>
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*
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* \note http://www.rowetel.com/codec2.html
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*
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* \ingroup codecs
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*/
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/*** MODULEINFO
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<depend>codec2</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include "asterisk/codec.h" /* for AST_MEDIA_TYPE_AUDIO */
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#include "asterisk/frame.h" /* for ast_frame */
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#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT, etc */
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#include "asterisk/logger.h" /* for ast_log, etc */
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h" /* ast_rtp_engine_(un)load_format */
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#include "asterisk/translate.h" /* for ast_trans_pvt, etc */
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#include <codec2/codec2.h>
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#define BUFFER_SAMPLES 8000
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#define CODEC2_SAMPLES 160 /* consider codec2_samples_per_frame(.) */
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#define CODEC2_FRAME_LEN 6 /* consider codec2_bits_per_frame(.) */
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/* Sample frame data */
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#include "asterisk/slin.h"
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#include "ex_codec2.h"
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struct codec2_translator_pvt {
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struct CODEC2 *state; /* May be encoder or decoder */
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int16_t buf[BUFFER_SAMPLES];
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};
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static int codec2_new(struct ast_trans_pvt *pvt)
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{
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struct codec2_translator_pvt *tmp = pvt->pvt;
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tmp->state = codec2_create(CODEC2_MODE_2400);
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if (!tmp->state) {
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ast_log(LOG_ERROR, "Error creating Codec 2 conversion\n");
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return -1;
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}
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return 0;
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}
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/*! \brief decode and store in outbuf. */
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static int codec2tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct codec2_translator_pvt *tmp = pvt->pvt;
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int x;
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for (x = 0; x < f->datalen; x += CODEC2_FRAME_LEN) {
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unsigned char *src = f->data.ptr + x;
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int16_t *dst = pvt->outbuf.i16 + pvt->samples;
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codec2_decode(tmp->state, dst, src);
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pvt->samples += CODEC2_SAMPLES;
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pvt->datalen += CODEC2_SAMPLES * 2;
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}
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return 0;
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}
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/*! \brief store samples into working buffer for later decode */
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static int lintocodec2_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct codec2_translator_pvt *tmp = pvt->pvt;
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memcpy(tmp->buf + pvt->samples, f->data.ptr, f->datalen);
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pvt->samples += f->samples;
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return 0;
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}
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/*! \brief encode and produce a frame */
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static struct ast_frame *lintocodec2_frameout(struct ast_trans_pvt *pvt)
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{
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struct codec2_translator_pvt *tmp = pvt->pvt;
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struct ast_frame *result = NULL;
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struct ast_frame *last = NULL;
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int samples = 0; /* output samples */
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while (pvt->samples >= CODEC2_SAMPLES) {
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struct ast_frame *current;
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/* Encode a frame of data */
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codec2_encode(tmp->state, pvt->outbuf.uc, tmp->buf + samples);
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samples += CODEC2_SAMPLES;
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pvt->samples -= CODEC2_SAMPLES;
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current = ast_trans_frameout(pvt, CODEC2_FRAME_LEN, CODEC2_SAMPLES);
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if (!current) {
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continue;
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} else if (last) {
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AST_LIST_NEXT(last, frame_list) = current;
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} else {
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result = current;
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}
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last = current;
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}
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/* Move the data at the end of the buffer to the front */
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if (samples) {
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memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
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}
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return result;
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}
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static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
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{
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struct codec2_translator_pvt *tmp = pvt->pvt;
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if (tmp->state) {
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codec2_destroy(tmp->state);
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}
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}
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static struct ast_translator codec2tolin = {
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.name = "codec2tolin",
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.src_codec = {
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.name = "codec2",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.dst_codec = {
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.format = "slin",
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.newpvt = codec2_new,
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.framein = codec2tolin_framein,
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.destroy = codec2_destroy_stuff,
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.sample = codec2_sample,
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.desc_size = sizeof(struct codec2_translator_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES * 2,
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};
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static struct ast_translator lintocodec2 = {
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.name = "lintocodec2",
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.src_codec = {
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.dst_codec = {
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.name = "codec2",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.format = "codec2",
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.newpvt = codec2_new,
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.framein = lintocodec2_framein,
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.frameout = lintocodec2_frameout,
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.destroy = codec2_destroy_stuff,
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.sample = slin8_sample,
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.desc_size = sizeof(struct codec2_translator_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1) / CODEC2_SAMPLES,
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};
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static int unload_module(void)
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{
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int res = 0;
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res |= ast_rtp_engine_unload_format(ast_format_codec2);
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res |= ast_unregister_translator(&lintocodec2);
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res |= ast_unregister_translator(&codec2tolin);
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return res;
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}
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static int load_module(void)
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{
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int res = 0;
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res |= ast_register_translator(&codec2tolin);
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res |= ast_register_translator(&lintocodec2);
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res |= ast_rtp_engine_load_format(ast_format_codec2);
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if (res) {
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unload_module();
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return AST_MODULE_LOAD_FAILURE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Codec 2 Coder/Decoder");
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32
codecs/ex_codec2.h
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32
codecs/ex_codec2.h
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@@ -0,0 +1,32 @@
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/*! \file
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* \brief 8-bit raw data
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*
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* Copyright (C) 2016, Alexander Traud
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*
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* Distributed under the terms of the GNU General Public License
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*
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*/
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#include "asterisk/format_cache.h" /* for ast_format_codec2 */
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#include "asterisk/frame.h" /* for ast_frame, etc */
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static uint8_t ex_codec2[] = {
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0xea, 0xca, 0x14, 0x85, 0x91, 0x78,
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};
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static struct ast_frame *codec2_sample(void)
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{
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static struct ast_frame f = {
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.frametype = AST_FRAME_VOICE,
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.datalen = sizeof(ex_codec2),
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.samples = CODEC2_SAMPLES,
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.mallocd = 0,
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.offset = 0,
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.src = __PRETTY_FUNCTION__,
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.data.ptr = ex_codec2,
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};
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f.subclass.format = ast_format_codec2;
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return &f;
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}
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